The professional quality (i.e., program) audio that is produced in a studio for a commercial radio broadcast must often be relayed to a remote transmitter site for over-the-air transmission. One technique known in the art for relaying such program audio over significant distances is to sample the audio to produce a digital data stream, then transmit the digital data on a TDM link such as a T1 or E1 digital circuit in the Public Switched Telephone Network (PSTN), Microwave links and other media. In order to preserve the integrity of the program audio, the sampled data is often transmitted in a linear range, i.e., not compressed in any way.
A common format for the digital audio data is the Audio Engineering Society/European Broadcast Union (AES/EBU) digital audio standard. AES/EBU is a bit-serial communications protocol for transmitting digital audio data through a single transmission line. This standard allows for two channels of audio data, up to 24 bits per sample, channel status bits for communication control and status information, and some error detection capabilities. Clocking information (i.e., sample rate), embedded in the AES/EBU bit stream, is recovered at the receiving end of the transmission path. The AES/EBU standard specifies the use of 32 kHz, 44.1 kHz, or 48 kHz sample rates.
FIG. 1 shows a prior art system for transmitting program audio in AES/EBU format from a studio site 12 to a remote transmitter site 14. A receive buffer 16 receives the AES/EBU data from the production source, the recovered clock and data are then provided to an asynchronous sample rate converter (ASRC) 18. The ASRC 18 converts the timing of the input AES/EBU data to conform to the TDM backplane clock associated with the studio multiplexor. The ASRC provides the data to the CPU 20 timed to the backplane clock. The PLL 24 derives the ASRC output clock, as well as the clock to the CPU 20, from the TDM backplane clock. The CPU 20 processes the AES/EBU audio data and passes this audio data to a common module (CM) 22, which implements (among other things) the T1 TDM functionality at the studio site 12. The CM 22 encapsulates the AES/EBU audio data into the T1 TDM link destined for the remote transmitter site 14.
A second CM 26 at the remote transmitter site 14 receives the T1 data from the studio site 12, extracts the timing from the T1 signal and the AES/EBU audio data from the T1 time slots, and passes the AES/EBU data to the CPU 28. The PLL 32 provides a clock to the CPU 28 that is synchronized to the TDM backplane clock. The CPU 28, after suitable processing, provides the AES/EBU data to a transmit buffer 30, which conditions and drives the AES/EBU audio to the appropriate destination within the transmitter site 14.
In the system of FIG. 1, all components after the ASRC 18 in the transmission path are synchronized to the TDM clock, so the audio data transfer from the studio site 12 to the transmitter site 14 occurs at a fairly constant rate. The ASRC 18, however, while ensuring a relatively smooth data transfer, can deteriorate the quality of the audio being transferred. For example, the ASRC FIR filter is a sharp-cutoff low-pass filter, which can produce “ringing” in waveforms that contain fast transitions. This can cause the peak FM deviation to be exceeded. Further, the finite arithmetic representation of the samples within the FIR algorithm requires one or more requantization steps, which introduce additional noise into the system. The additional hardware necessary to implement the ASRC 18 adds to the overall cost of the system. The program audio is typically relayed to the transmitter in a linear range, i.e., without compression, to avoid similar deterioration in the audio quality. It is therefore undesirable to have the ASRC 18 in the data path.